I
have recently been asked by a well-known broadcasting organist to
undertake a recording of a Wurlitzer pipe organ in an auditorium. My
first thought would be to use a spaced pair, or perhaps an ORTF pair,
far back in the auditorium to take in the acoustics, and maybe another
mic or two next to the pipe chambers for the definition of the
percussion and so on. But I'm not sure about distances of mics or
accounting for the delays between them. I'd like to know if I'm in the
right ball park, or a long way off.

The multi-capsule Soundfield mic is a popular choice for all kinds of location recording.

Via Email

Technical Editor Hugh Robjohns replies: As I'm sure
you know, no two Wurlitzers are ever exactly the same, and certainly no
two halls are the same, so I'm afraid I can't offer any exact
information for you. It really is a case of going by ear.

The ORTF approach is generally a good one, although a lot of people
like to record organs using true coincident pairs. The multi-capsule
Soundfield mic is also a common favourite for this kind of job.
As to distance, it all depends on the acoustics of the hall (in other
words, the Critical Distance) and what kind of perspective you require
in the recording, which may change with each item performed, of course.
The Critical Distance (abbreviated to Dc) is the point, measured from
the sound source, at which the direct signal and the reflected or
reverberant signal are of equal intensity.

To find the Critical Distance, you'll need to use a long tape
measure, a sound level meter and some means of generating a reasonably
constant level of sound in the room. It doesn't need to be at PA levels —
something approaching the spoken voice will be fine. You could use a
radio tuned to a heavily compressed commercial pop music radio station,
for example, but a decent active speaker and a source of pink noise
would be better.

Start by measuring the noise level about 10cm in front of the speaker
and make a note of the reading. Then double the distance to 20cm and
measure again. The noise level will have reduced by something between 4
and 6dB because we are in the direct field where a doubling of distance
results in a halving of level. You then keep doubling the distance and
measuring the drop in level from the previous position, so measure the
level at 40cm, 80cm, 160cm, 320cm and so on. While you are within the
direct field, each doubling of distance results in a level drop of
between 4 and 6dB, but as you near the point where the direct and
reverberant fields are equal in level, the level drop will get much
smaller — a change of only a decibel or two indicates that you have
found the Critical Distance. Further increases in distance will result
in no significant change in level at all, because you are now in the
reverberant field.

As you move away from the source, the closer to the Critical Distance
you place the mic(s), the more reverberation you will pick up. Moving
the mic(s) closer to the source will result in a drier signal. Normally,
an ORTF array using cardioid mics would need to be placed at roughly
Dc/2 (half the Critical Distance), and a spaced pair of omnidirectional
mics would be placed at something like Dc/3 (one-third the Critical
Distance).

In general, organ lofts and pipework are built fairly high, and so a
very tall mic stand or two help to get the mic(s) on axis to the
pipework. It's then a case of moving the mic around to get the best
balance you can between the different sections. If the pipework is
installed in multiple locations, you may need to use several mics to
cover everything to attain the best balance. In this case, you may well
be able to achieve an acceptable stereo effect using separate, panned,
mono mics instead of (or in addition to) a stereo pair.

Personally, I tend to go for a main stereo pair to give the best
overall balance and acoustic impression of the room, and then add
additional (usually) mono mics if needed to reinforce a particular
section, just to provide a little extra clarity or definition.

Friday, October 28, 2016

Magix Samplitude offers some unusual and attractive features, like the Elastic Audio editor.

I've been using a Roland VS1680 for about six years now, and once
I've finished off a couple of on-going projects I'm looking to upgrade
to a PC setup. Until I read Paul Sellars' Magix Samplitude Professional
v8 review in SOS June 2005, I was homing in on a specialist PC,
Steinberg Cubase SX, probably a TC Powercore, some Waves plug-ins, and a
Yamaha 01X front end, but I'm now also considering the Mackie Control
as an alternative to the 01X. I know this is a difficult one, but can
you tell me what you think would be best?

Dave Gornall

Features Editor Sam Inglis replies: It's tough to
decide between competing programs such as Cubase SX and Samplitude —
very probably, both of them will have all the features you need, and so
it's a question of finding which one best suits your chosen way of
working. For that reason, nothing beats being able to try them out for
yourself, so if you know anyone who's using those packages, try to get
some time with them before you make your decision. Failing that, see if
you can arrange an in-depth demonstration at a local music store.

As I understand it, the most important difference between the two is
that Cubase SX works on a traditional 'virtual studio' model,
where audio lives on tracks and effects are applied on mixer channels.
Samplitude can work this way too, but its Arrange page is designed so
that each chunk of recorded audio is treated as an independent Object,
with automation and effects settings applicable to individual Objects
rather than mixer channels or tracks. Some people find this way of
working more to their taste, especially for mastering applications.

As for the Mackie Control versus 01X debate, one thing you'll need to
consider is that the Mackie Control has no audio I/O, so if you take
that route you will need to budget for a separate audio interface.

Tuesday, October 25, 2016

I'm
about to move house and set up a studio in the spare bedroom with all
my stuff, and I'm just interested in what the best use of power points
is. Is it acceptable (and safe!) to run many plug boards (between 24 and
28 at least) from one outlet in the wall, or is it better to use all
the available outlets around the room? I have quite a lot of stuff to
power in one small room, and I am thinking about perhaps getting a power
conditioner as well — I don't want to be responsible for burning down
our landlord's house!If
you want to avoid ground loops, it's best to run all your music-making
gear from a single mains socket (like this UK one, shown).

SOS Forum Post

Technical Editor Hugh Robjohns replies: From the
point of view of avoiding ground loops, it is best to run everything
from a single socket, or from adjacent sockets if you have a
double-socket outlet. This is a much better approach than running some
gear from a socket on one side of the room, and other gear from a socket
on the other side — a practice almost guaranteed to produce ground-loop
problems!
If you are concerned about the total power you will be drawing from a
single socket, you can reduce the load by plugging non-audio equipment
into another socket in the room — desk lamps, phone chargers, kettles
and so on.

If you're using plug boards to increase the number of available
sockets, connect them in a star arrangement rather than serially. By
that I mean you should connect one board to the wall socket, then plug
the other boards into that first one, and then plug the equipment into
this second 'layer' of boards. That way, the earth paths are kept as
short as possible and in a star arrangement.

A single socket is able to supply around 3kW (230V x 13 Amps) and it
is very unlikely that your domestic recording equipment will draw that
much power — but every piece of equipment will have a label on it near
the power connection that says what power (or current) it draws. If
possible, try to balance the power demands on each plug board and make
sure that all the fuses (in the plug boards and in individual plugs) are
sensibly rated. Also, bear in mind that the first plug board has to
carry the entire current load.

Saturday, October 22, 2016

When sync'ing external MIDI gear, the Yamaha AW16G can act as the MIDI Clock master but not as the slave.

I have a song recorded as a MIDI sequence on a Yamaha QY700 hardware
sequencer which I'm trying to record onto a Yamaha AW16G. I have the
QY700 slaved to the AW16G using MIDI Clock, so that when I start and
stop recording on the AW16G, the QY700 starts and stops in sync. This
works well except towards the end of the song, where there is a slowdown
in tempo programmed into the sequence on the QY700. At this point, the
tempo display on the QY700 indicates that the ritardando occurs as it is
meant to, but the AW16G continues recording at a steady 140bpm. I've
tried slaving the AW16G to the QY700 using MIDI Clock, but cannot seem
to get this to work. I've also tried setting the QY700's MIDI ports to
MTC, with the AW16G set as an MMC slave, but this does not work either.
Help!

Stuart Tatlock

Reviews Editor Mike Senior replies: The reason why
the AW16G doesn't follow the tempo change programmed into the QY700 is
that the QY700 is slaved to the AW16G, and not the other way round. MIDI
Clock messages from the master machine fix the rate of quarter notes
according to tempo settings on the master machine, not the slave. What
you need to do is program any time-signature and tempo changes into the
AW16G's tempo list — the manual will tell you how to do this. Once this
is done, the tempo of the master machine will slow down towards the end
of the track, and the tempo of the slave machine should then follow it.

Referring back to the original review of the AW16G in SOS October 2002 (www.soundonsound.com/sos/Oct02/articles/yamahaaw16g.asp), it seems that the multitracker can only act as the MIDI Clock master and
not the slave, which explains your difficulties in trying to achieve
this! It's not uncommon to find that less expensive multitrackers can
only act as the master, but it's much better to work with the sequencer
as the slave anyway, so don't worry about this.
The AW16G can however work as the master or the slave in the case of
MTC and MMC data, if you wanted to take that route, though getting MTC
to work is a little fiddly on the QY700. Also, it's probably worth
mentioning that MTC and MMC are different things. MTC is MIDI Time Code,
which is a timecode just like SMPTE. MMC is MIDI Machine Control, and
this is a set of remote transport-control functions — things like Play,
Stop, Fast Forward and so on. The two are often used together, and
you'll have to use both to get the QY700 and AW16G sync'ing properly,
but the protocols shouldn't be confused.

As an aside, if you slaved the QY700 to the AW16G using MTC, then the
sequencer's tempo changes would remain intact on playback. However, the
bars/beats displays on the two machines still wouldn't match up unless
you programmed the AW16G's tempo list to match that of the QY700. This
is because MIDI Time Code works in terms of hours, seconds, and minutes,
not bars and beats. If your tempo changes on the QY700 are ridiculously
involved, so you don't want to have to transfer them over to the AW16G,
you might prefer to synchronise using MTC, if possible, and put up with
the erroneous bars/beats display on the AW16G instead.

Thursday, October 20, 2016

Dedicated
effects processing hardware, such as the TC Powercore Compact, offers a
way to make use of the latest reverb plug-ins without placing
additional demands on the host computer's CPU.

How much difference is there between the quality of a VST reverb
plug-in and a hardware reverb processor such as the Lexicon PCM81 or the
TC Electronic M One XL?

SOS Forum Post

Technical Editor Hugh Robjohns replies: Digital
reverb effects, or at least reverb effects that try to emulate real
spaces with some degree of accuracy, involve a great deal of complex
digital data processing. While it is perfectly possible to run the
algorithms in a PC or Mac environment using the host processor, the
maths involved places huge demands on the CPU. As a result, even on the
most up-to-date and powerful systems there will be a practical limit to
the number and complexity of reverb processors that can be run while
doing everything else you may want the computer to do.
Hardware reverbs are dedicated to doing just one thing, and so can be
heavily optimised in terms of the processing power and electronic
design.

Up until recently, most native reverb plug-ins used relatively simple
algorithms and were often audibly inferior to even quite modest
hardware reverbs. The advent of software convolution has improved
matters considerably, and many of these new convolution reverb plug-ins
sound as good as hardware units in many situations (to my ears at
least).

Another very good alternative is to use embedded hardware processing
like the TC Powercore or Universal Audio UAD1 cards. These offer
dedicated DSP power to avoid clogging up the host processor, and allow
advanced algorithms (often transcoded from hardware processors) to be
run. The advantage is that everything is still under computer control,
and so settings can be instantly saved with specific projects, which
makes remixing or revising a project later on much quicker and easier
than trying to find your settings notes (if you remembered to write them
down!) and/or reconfiguring a hardware unit.

Wednesday, October 19, 2016

The Roland VS2400CD's flexible routing system can seem complicated at first.

Please can you help me understand the effects routing on my Roland
VS2400CD? Is it possible to have effects on all tracks with only one
effects card installed? Can you explain (in idiot's terms) how to
'print' an effect to a track? Also, would this free up the effect for
use on other tracks?

Helen Adams

Reviews Editor Mike Senior replies: First of all,
let's clarify some things about setting up effects routing in general,
regardless of which system or machine you're using. Basically, there are
two ways you can add effects to a single track. The simplest is to just
shove the effect directly into the signal path, something referred to
as 'inserting' the effect. This is the best approach for any of the
algorithms which invlove compression, and anything which is designed to
change the character of the whole signal, such as track-specific EQ
treatments and modelling or modulation processors.
The disadvantage of using your onboard effects as inserts is that you
need a separate effects processor for each track you want to add
effects to. It is to get around this problem that you're given
compression and equalisation as standard on every channel — if you had
to add these using the effects board, one board wouldn't get you very
far!

Setting up a send effect is the best way to use an effect on multiple tracks.

The second routing approach allows you to use your choice of effect
algorithm on all the tracks at once. This is done by feeding the
effect's FX Return channel from one of your aux sends, such that any and
all tracks can send to the effect using the channel aux send controls.
You then return the output of the effect to the mixer for mixing
alongside your tracks. This is the best approach for any effect which
makes use of any kind of delay or reverb.

I'm guessing that you've already worked out how to insert effects, as
that's much easier, so here's how to set up a send effect. Let's assume
you're wanting to add reverb to every track. First push the
Aux1-8/FX1-8 button to bring up the FX Return channels to the faders,
and then press the Ch Edit button to bring up the settings for the FX1
Return. There will be a little box at the top left of the screen
labelled Assign. Set this to a spare auxiliary send — for the purposes
of this example, let's use Aux 1. While you're in that screen, check
that the fader is up, the Mute switch is off, the Mix switch is on, and
the Aux 1 switch is off (the last of these avoids the possibility of
creating a feedback loop).

Now cursor to the large Effect 1 box and press the Enter button
(below the main data wheel). This will take you to the screens where you
can select the effect and change the parameters if you wish. You can
also reach these effects-editing screens at any time by holding Shift
and pressing F4. Now choose a channel to which you want to add effects,
and press its Ch Edit button so that you can see its channel parameters.
Along the bottom of the screen you'll see all the aux send controls.
Cursor to Aux 1 and switch the switch to Pst. This setting means that
the signal is sent to the effect from after the channel's level fader
(post-fader, hence 'Pst'), which is the best choice in this case as it
means that the effects level you set shouldn't need adjusting if you
change the fader level. The other setting, Pre, is best when you're
using the Aux send for other purposes, such as for providing monitor
mixes during recording.

Setting the channel's aux send to 'Pst' (post-fader) is best when using send effects.

Now cursor to the Aux 1 control and use the data wheel to fade it up.
You should now (fingers crossed) hear the reverb effect being added. If
you don't, then you should try to troubleshoot the situation by using
the available metering. Go to the Home screen, where a row of meters are
given at the top of the screen. The 'F' keys at the bottom let you show
a variety of different signals with these meters. In this case, you
could have a look at the aux buss meters to check that signal is
reaching Aux 1 from the channel, and have a look at the FX Return meters
to check whether signal is reaching the effect channel. The
ToPre/ToPost option on the F6 key is particularly useful. If you have
the metering set to Pre (the current setting is shown above the meters)
you can see whether a signal is reaching the input of a given channel,
whereas if you have the metering set to Pst you can tell whether it's
leaving a given channel. For example, if the metering shows a signal for
FX1 Return in the Pre metering mode, but not in the Pst metering mode,
then you know that there's some channel setting that's not letting the
signal through the FX1 Return channel — perhaps the Mute button is on,
or the Mix switch is off. Learning how to troubleshoot routing problems
using the metering is incredibly useful on the VS multitrackers, so do
take the time to learn about how the metering works, if nothing else. It
may seem a bit boring, but it really pays off when things don't go how
you expect them to!

Now let's turn to printing effects. Depending on whether you're using
insert or send effects, the procedure for printing effects is
different. However, for both, you'll need to use the internal routing
matrix. First choose the track you want to print to, and then hold down
its Status button for a second or so until the patchbay screen comes up.
You can also access this screen from the EZ Routing button. Hold down
the Status button again and press Clear to remove any existing
assignments to that track, and then press the Ch Edit button on the
source track. A little line should now show up on the screen connecting
the source channel on the Track Mixer block with the destination channel
on the Recording Track block.

Now go back to the Home screen, arm the destination track for
recording, and print the track to its new destination with its effects.
If you need to, you can print a mix of several tracks, along with the
output of an effects return, in the same way — just make sure that all
the channels' and returns' Ch Edit buttons are lit at the relevant
point. You can even bounce to a stereo track if you've got one set up.

Monday, October 17, 2016

I
have a problem that I think could be fairly widespread. I've spent
years dabbling with studio technology, reading SOS, acquiring various
bits of kit, messing around with software and so on. The trouble is,
when I do get a few spare minutes to actually sit down and create music I
seem to be devoid of inspiration — I've spent so long worrying about my
next bit of kit I seem to have lost sight of the object, which is to
make music. Help! I need something to kick-start my creativity, but I
don't know what it is. Can you offer any advice?

Neale, Cambridge

Editor In Chief Paul White replies: This is a common
problem, but it helps if you can divide your time between music-making
and and the more mundane tasks of studio maintenance and management.
There's nothing worse than having to take your mind off composing to
deal with a computer issue. If you don't feel inspired, then tidy your
hard drive, update your plug-ins, organise your files and so on. The
other useful thing you can do is construct a default song with all your
commonly used software instruments ready to go so that as soon as your
computer boots up, you can open your default song and start work. If you
can switch on your computer and then leave it in Sleep mode when you
have spare time, it will make starting work that much faster.

A dictaphone or a portable recorder like the

M-Audio MicroTrack can be handy for

capturing new ideas when inspiration strikes.

Nothing kills creativity like spending an hour looking for sounds, so
try to keep your sound library organised, and if you do embark upon a
sound browsing session between creative bursts, try to put your
favourites into suitable categories for future use. Having a million
samples is of no use if they are not organised. The better you prepare,
the quicker you can get your ideas down when they arrive. I also find
that having a good rhythm part often inspires ideas and that's where
something like Stylus RMX is excellent. Even if you decide to
replace or change the rhythm parts later, it is worth using a preset as a
starting point if it gives you some ideas. Some composers also find
that setting artificial limits gets them moving faster, so why not give
yourself a dozen sounds to choose from, and see what you can create
using them? Once the idea takes shape, you can break this rule and
proceed as normal.

Other useful tips include keeping a simple dictaphone with you so
that when an idea pops into your head halfway down the M4, you can hum
it or sing it and then come back to it later. Strictly speaking, this
should have a hands-free mic attached! It can also help to turn your
back on the technology for a while and then just sit down with a guitar
or piano and noodle around for a while. Again, leaving a
portable recorder running is a good idea, just in case anything
wonderful and totally unrepeatable should turn up! And finally, some
people find it much easier to come up with ideas when they have a
musical colleague to bounce their ideas off, so don't always assume that
you can do everything alone. Different things work for different people
— for me, having a deadline definitely does the trick!

Friday, October 14, 2016

I've
just bought my first mic, a Rode NT1A, which requires phantom power,
and I'm wondering if there's anything wrong with leaving it plugged in
with phantom power switched on for long periods of time? It's a bit of a
hassle to have to take it off the stand, unplug it and put it away
every time I stop using it.

SOS Forum Post

Technical Editor Hugh Robjohns replies: There's no
problem with leaving your mic powered up — many professional studios
prefer to keep their mics powered all the time, and there are some good
arguments in favour of this approach.

Leaving
a condenser mic such as this Rode NT1A on it's stand and powered up
when not in use does no damage, though you might want to keep the dust
off with a polythene bag.

The most likely cause of damage to a mic is accidentally dropping it
on the floor, and that is most likely to happen when putting it on or
taking it off a stand! So avoiding that risk is probably a good thing.
Also, if the mic is supported in a shockmount, continually taking the
mic in and out of the mount will tend to stretch the suspension elastics
and weaken the clamps, again making it less reliable over time.

So, assuming that you have space to leave your mic on its stand, it's
not a bad idea. However, you should take steps to make sure that the
capsule is protected from dust when not in use, so placing a clean
polythene bag (a freezer bag, for example) over the mic when you have
finished with it is a sensible precaution.

Leaving the mic powered will also help to keep moisture at bay in
most locations (as long as we are talking about a room in a house rather
than a shed in the garden). As well as keeping dust away, a polythene
bag over the mic will also help keep warm air around the capsule.
Most electronic components are quite happy if powered permanently,
and generally fail when power is applied after being turned off. So
leaving a mic powered is unlikely to shorten its life significantly, and
may well actually prolong it.

As I said, a lot of high-end studios leave their mics on stands,
powered and protected with bags — it's not an unusual practice at all.
Of course, in those situations the mics are in use pretty much every
day, but if having the mic on a stand, powered and ready to go, helps
make it easier to record something when the mood takes you, why not? It
would be a shame not to record just because you can't be bothered to get
out a mic stand, unpack a mic and plug it all up!

Saturday, October 8, 2016

Is
the German GEMA essentially the same as the MCPS? Does a band putting
out its own CDs need to register with different people in different
countries, or do these organisations cover all situations?

Via Email

SOS contributor Tom Flint replies: GEMA performs
pretty much the same function in Germany as the MCPS (Mechanical
Copyright Protection Society) does in the UK. GEMA's full name
(Gesellschaft für musikalische Aufführungs- und mechanische
Vervielfältigungsrechte) translates as the Society for Musical
Performing and Mechanical Reproduction rights. In other words, GEMA help
songwriters, lyricists and music publishers obtain their royalties and,
just like the MCPS, GEMA acquires these funds by taking a cut of record
sales revenue in exchange for granting manufacturing licences to record
labels.

Releasing a record commercially requires a fair amount of paperwork.

In the UK, the MCPS licences usually have to be paid by the record
label up front and are set at 8.5 percent of the price the label charges
the distributor for each record (known as the PPD or Published Price to
Dealer). The 8.5 percent is the writer's cut of the record's sale
price, although writers who are signed to a publisher have to split
their fee according to their publishing deal. If no dealer or
distributor is involved, the figure paid by the record label is rated at
6.5 percent of the retail price, excluding VAT. GEMA operate in a
similar way, although they take just over 9 percent of the PPD.

Other countries besides Germany also have their own versions of GEMA.
In France, for example, there is SACEM, in Japan JASRAC, and in the US
they have the Harry Fox Agency.
Quite whether you will actually need to deal with GEMA, or any other
foreign agency depends on your location. According to the MCPS,
licensing is not determined by the country of manufacture, but by the
country in which the label is based. This means that if you are a
UK-registered company it won't be necessary for you to get a licence
from GEMA, even if you are using a German manufacturing company to make
your CDs. The same is true if you are manufacturing CDs in the UK and
exporting them to Germany. Obviously you could strike some sort of deal
with a German label and have them release the record on your behalf, but
it would then be up to them to obtain the relevant licence from GEMA.

It's worth noting that the MCPS are not the only collection society
you need to consider contacting when releasing a record. There is also
Phonographic Performance Limited (PPL), which collects licence fees for
records played on the radio and TV and in pubs, clubs and other public
places, and the Performing Right Society (PRS), which collects royalties
from the public performance and broadcast of musical works (both
recordings and live performances). Fortunately, both the PPL and PRS
gather musical performance royalties from foreign countries on your
behalf, so you don't necessarily have to sign up to the equivalent
organisation in each and every country.

Thursday, October 6, 2016

I
recently purchased a second-hand Tandberg reel-to-reel tape machine and
I'm having difficulties connecting it to my external hi-fi. I was
provided with a lead that has a five-pin socket at one end and phono
leads at the other, which I plug into the 'analogue in' socket on my
hi-fi. However, when I'm playing tapes the music only comes out of one
channel. The back of the Tandberg has two of these five-pin sockets and
also three other holes, marked 'p up', 'amp' and 'radio'. Can you tell
me how I can get the sound coming from both speakers and not just one?
Any help would be most appreciated by this novice reel-to-reel owner!

SOS Forum Post

Technical Editor Hugh Robjohns replies: There are
several possibilities here. The most obvious one is that the DIN-phono
lead you have is broken. DIN is the Deutsches Insitut für Normung, a
German standards-setting organisation, and it specified a range of
connectors using a similar body with between three and 14 pins. The
three- and five-pin versions were used a lot on hi-fi equipment in the
'60s and '70s, before the RCA 'phono' socket became the standard
interface, and now the five-pin DIN is most commonly found on MIDI
leads. If you have a test meter, check the connections between the phono plugs and DIN pins to see if the cable is faulty.The 'standard' numbering scheme for DIN plugs.

For some bizarre reason, some manufacturers' implementation of the
DIN wiring is exactly the opposite of others, so although I am giving
the most common way of wiring them up, bear in mind that this is not
always the case. The 5-pin DIN sockets were used to convey stereo
unbalanced signals. The DIN pins on a male jack are numbered in the
order 1, 4, 2, 5, 3, clockwise from right to left (see diagram).
Normally, pins 1 and 4 were used for the left and right inputs,
respectively, and 3 and 5 for left and right outputs, with the middle
pin of the five (pin 2) serving as the common screen or earth connection
for all four signals. If your DIN-phono lead only has two phono
connectors on it, the centre pins of the two phonos will either go to 1
and 4, or 3 and 5 — a test meter will help you find out which.
The other possible explanations for why you're only getting output on
one channel are broken electronics within the machine itself, or that
you are trying to play a quarter-track tape on a half-track machine (or
vice versa)...

You can check the latter by looking at the heads or making a test
recording to a blank tape. A half-track head uses almost half the tape
width for each channel, so you'll see the two head gaps occupying just
under half the tape width, with only a small gap (guard band) between
them. A quarter-track head uses slightly less than a quarter of the tape
width for each track, and the two channels are separated by a
quarter-track width, so the two head gaps are separated by the width of
another head gap.
As for the 'p up', 'amp' and 'radio' sockets, this suggests that the
machine has a built-in record selector and preamp. 'P Up' will be an
RIAA phono pickup input, for example. 'Radio' is pretty
self-explanatory, and 'Amp' is probably another line-level input — but
it could possibly be an output intended to go to a preamp. It would be
worth checking anyway!

Wednesday, October 5, 2016

A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.

I never did quite understand the subtle differences between all the
different variants of Dolby — A, B, C, HX and SR. Could you explain them
to me? Are there any others I've missed? What are Dolby Labs doing
these days? I guess they've undergone some 'reduction' themselves...

SOS Forum Post

Technical Editor Hugh Robjohns replies: Dolby A was
the first professional noise-reduction system — launched in 1967 if
memory serves — and it used four separate frequency processing bands.
You can think of them crudely as bass, mid-range, treble and high
treble, with the top two overlapping so that the 'hiss region' was
processed more heavily than the rest. Avoiding line-up errors between
encoding and decoding was crucial, so the infamous Dolby warble tone was
used to identify encoded tapes and to allow accurate replay alignment.
Dolby A was originally used to get respectable audio performance out of
early professional video recorders, but was later adopted for multitrack
recording and cinema optical soundtracks.

Dolby B was a very simple domestic system intended to improve the
performance of compact cassette recorders. It was also used on some
later domestic quarter-inch machines. Dolby B was a single-band system
affecting only the high end, with very modest compansion. It had no
facility, or indeed any practical need, for replay alignment.

Dolby C was a much more aggressive multi-band version originally
intended for small-format professional video-tape systems and
narrow-gauge semi-professional studio multitrack recorders. It was very
sensitive to mistracking, but was unfortunately designed without any
line-up tone facility to calibrate playback levels.

In the professional market, Dolby A was superseded by Dolby SR, which
was Dolby's most sophisticated multi-band noise reduction system. This
employed 10 bands altogether, some operating at fixed frequencies and
others moving automatically to suit the material, and allowed the user
to achieve a signal-to-noise ratio of around 90dB from analogue tape.
However, although it was a very clever and effective system it arrived
just a few years too late and the digital revolution effectively
eclipsed it. Dolby SR used a modulated noise signal for identification
and replay alignment.

Finally, Dolby S (one you missed off your list) was a last-ditch
attempt aimed at semi-pro and domestic recorders, and was a halfway
house between Dolby SR and Dolby C. It still had no built-in line-up
facility, though. It was used on some semi-pro narrow-gauge
multitrackers and the last of the high-end hi-fi cassette recorders.

Dolby HX is not a noise-reduction system at all — it is a clever
system to avoid over-biasing on analogue tape machines using high-output
tapes. This system was used on some high-end domestic cassette
recorders and the last of the professional analogue two-track machines,
such as the Studer A807. Dolby HX is a once-only process that needs no
decoding. In essence, it reduces the bias level if there is a lot of
high-frequency content in the audio signal, thus preventing over-biasing
and the noise artefacts and frequency-response errors that go with it.

Dolby Labs still make Dolby SR and A systems for analogue multitrack
and cinema applications, and I guess they are still collecting licensing
revenues from the other systems when they are used on domestic cassette
recorders and the like. However, most of the company's efforts these
days are geared towards digital data-reduction systems, which are based
entirely on the frequency-masking principles first exploited by Dolby's
analogue noise-reduction systems. That is why Dolby AC3 has always been
amongst the best of the data-reduction codecs for a given data rate —
the company had a major head start on the rest of the field.

Tuesday, October 4, 2016

Kjaerhus Audio's Golden Unipressor allows the use of side-chain inputs in Cubase SX.

I've been using Steinberg Cubase SX for a couple of years now, but
although the bundled VST dynamics plug-ins in SX 2 are perfectly
reasonable, after reading up on various production methods I've realised
that professional studios make a lot of use of 'ducking' and
'side-chaining' compressors in order to keep the vocals or lead
instruments in balance with the main body of the mix, for example. I
know this sort of thing is possible in Reason or Live, but from what I
understand of Cubase, signals cannot be 'routed' in this way to a
compressor's side-chain to modulate another signal. Or could it be done
with something like the Xlutop Chainer or some such virtual router? The
Waves plug-ins feature side-chain options, but I wouldn't know how to
route a signal into them (and, as far as I know, Chainer doesn't support
them). I'd greatly appreciate your advice.

Dale Kunzler

Features Editor Sam Inglis replies: You're quite
correct that the SX mixer is not very good for side-chaining using
external key inputs. In fact this is true of many other DAW programs,
too; the only one I know that currently offers the necessary flexible
bussing structure is Pro Tools. For instance, I'm pretty sure that the
side-chain features on Waves plug-ins only support external key inputs
when used in Pro Tools — in SX and other applications, the side-chain is
always the same as the audio input itself.
Recently, however, a few plug-ins have appeared that allow you to
compress a stereo track in SX with the side-chain keyed from a separate
source track. The way this works is a bit clumsy: you need to create a
surround channel on the mixer, then route both the audio source and the
side-chain signal to it. That way, a 'surround' plug-in inserted on that
channel can 'see' both the audio source and the side-chain signal.
Plug-ins I'm aware of that allow you to do this include Otium FX's
Compadre, DB Audioware's dB-D dynamic processor and Kjaerhus Audio's
Golden Unipressor (shown above). I should point out that I haven't
tested any of them, though!

Monday, October 3, 2016

With
properly designed and constructed monitors, whether active or passive,
you needn't worry about internal vibrations damaging electrical
components.

I would like to know how much benefit can be gained by isolating my
speakers and other gear from their supports — so-called 'seismic
isolation'. I recently saw a forum posting suggesting that passive
monitors have an advantage over active ones as, in the case of the
latter, vibrations from the speaker can affect the components of the
built-in amp. The benefits of decoupling equipment using springs that
have a very low resonance frequency so that it 'floats' was also
discussed. Apparently many things can benefit from this technique — not
just speakers but CD players and studio gear also. Improvements to the
stereo field and depth are said to be quite noticeable. I would like to
know if the £1300 I spent on an active Blue Sky System One (which I like
very much) would have been better spent on passive speakers and an
external amplifier. Can you shed any light on all of this for me please?

SOS Forum Post

Technical Editor Hugh Robjohns replies: With regards
to what you read about internal vibrations in monitors, in general
neither the active or passive form has an advantage in this regard, and
both potentially suffer exactly the same problem.

Clearly, there is a lot of sound energy inside most loudspeaker
cabinets, and if that energy is allowed to impact on electronic circuit
boards it is possible that some components might resonate and vibrate,
eventually resulting in damage to the solder joints or the components
themselves, and possibly such mechanical resonances might affect the
electrical signal passing through the components. However, this would
apply equally to passive crossover boards as much as active amplifiers.

In 30-odd years of playing around with loudspeakers in many and
various forms, I can't say I have ever found this to be a real problem. I
have occasionally come across speakers that have suffered component or
solder joint failures, but in all cases the causes have been traced to
faulty production or failures in quality control. When the faults were
fixed properly, none recurred as far as I am aware — even though you
would expect them to if the sole cause was sound vibrations within the
cabinet. So I am confident that this argument can be set aside as a
popular but completely unfounded myth.

Mechanical isolation of speakers or other devices from their supports
can be used to advantage in certain situations, but it is a complex
subject and it is easy for the inexperienced to make the situation worse
with inappropriate decoupling systems. In my experience, most equipment
works best when mounted on solid, heavy supports — there is nothing as
effective at controlling vibrations as a lot of mass.
However, sometimes it is necessary to come up with some form of
decoupling to prevent vibrations generated in one source from entering
an adjacent surface. The classic example is that of placing nearfield
speakers on a desktop, when the inherent speaker cabinet vibrations will
often cause the desktop to vibrate and resonate, resulting in unwanted
rattles. In this situation, placing the speakers on some form of
decoupling medium can improve matters — something like the Auralex
Mo-Pads, for example, are very effective. However, far better results
can be obtained by removing the speakers away from the table top
completely and mounting them properly on solid, heavy stands placed
directly on the floor.

Auralex Mopads can be used to isolate monitors placed on a desktop, but heavy-duty floor stands are best of all.

As far as equipment is concerned, I don't subscribe to the view that
properly designed and manufactured amplifiers and other electronics
should be decoupled to improve stereo imaging or anything else. However,
when it comes to systems involving some mechanical element — like
record players, CD players and so forth — unwanted vibrations entering
the mechanical system certainly can cause problems.

Most people are very well aware of the susceptibility of record
players to external mechanical or acoustic vibration. The required
tracking precision in CD players and DVD players is many orders higher,
and mechanical vibrations that reach the mechanism will affect the
accuracy of the tracking. Potentially, this will cause the tracking and
focus servos to work harder, forcing greater current flows at higher
frequencies through the motors. In cheaper designs, this may well affect
the power supply's stability and result in noise currents reaching
other parts of the circuitry. Reduced tracking precision can also
potentially result in a greater uncorrected error rate and far more
jitter. Cheap and poorly designed players are likely to suffer these
effects to a much higher degree than properly engineered equipment,
which will usually incorporate properly decoupled drives, effective
de-jittering circuitry, and so on.

It's a familiar scenario in the hi-fi world — people discover that
badly engineered equipment reacts 'unexpectedly' to different cables,
mechanical decoupling, or painting with a green pen — all of which
bestow a 'miraculous' benefit to the sound... and then declare (from no
scientific basis whatever) that all vaguely similar equipment will
behave the same. It's just not the case.

As to whether you would have been better off buying passive monitors
and an amp, the answer is probably not. I can think of some excellent
passive monitor and amp combinations for the rough cost you mention, and
in direct comparisons I dare say some people would prefer a passive
speaker and amp configuration over your Blue Sky System One. But it
comes down to personal preferences regarding sound, convenience and
styling, and how the system works in a given room. I think you can
continue to enjoy your Blue Sky system and completely disregard any faux concerns raised by the technical myth-spreaders!